1. Field of the Invention
The present invention relates to acoustics, and, in particular, to microphone arrays.
2. Description of the Related Art
A microphone array-based audio system typically comprises two units: an arrangement of (a) two or more microphones (i.e., transducers that convert acoustic signals (i.e., sounds) into electrical audio signals) and (b) a beamformer that combines the audio signals generated by the microphones to form an auditory scene representative of at least a portion of the acoustic sound field. This combination enables picking up acoustic signals dependent on their direction of propagation. As such, microphone arrays are sometimes also referred to as spatial filters. Their advantage over conventional directional microphones, such as shotgun microphones, is their high flexibility due to the degrees of freedom offered by the plurality of microphones and the processing of the associated beamformer. The directional pattern of a microphone array can be varied over a wide range. This enables, for example, steering the look direction, adapting the pattern according to the actual acoustic situation, and/or zooming in to or out from an acoustic source. All this can be done by controlling the beamformer, which is typically implemented in software, such that no mechanical alteration of the microphone array is needed.
There are several standard microphone array geometries. The most common one is the linear array. Its advantage is its simplicity with respect to analysis and construction. Other geometries include planar arrays, random arrays, circular arrays, and spherical arrays. The spherical array has several advantages over the other geometries. The beampattern can be steered to any direction in three-dimensional (3-D) space, without changing the shape of the pattern. The spherical array also allows full 3D control of the beampattern.
Speech pick-up with high signal-to-noise ratio (SNR) is essential for many communication applications. In noisy environments, a common solution is based on farfield microphone array technology. However, for highly noise-contaminated environments, the achievable gain might not be sufficient. In these cases, a close-talking microphone may work better. Close-talking microphones, also known as noise-canceling microphones, exploit the nearfield effect of a close source and a differential microphone array, in which the frequency response of a differential microphone array to a nearfield source is substantially flat at low frequencies up to a cut-off frequency. On the other hand, the frequency response of a differential microphone array to a farfield source shows a high-pass behavior.
FIGS. 1(a) and 1(b) graphically show the normalized frequency response of a first-order differential microphone array over kd/2, where k is the wavenumber (which is equal to 2π/λ, where λ is wavelength) and d is the distance between the two microphones in the first-order differential array, for various distances and incidence angles, respectively, where an incidence angle of 0 degrees corresponds to an endfire orientation. All frequency responses are normalized to the sound pressure present at the center of the array. The thick curve in each figure corresponds to the farfield response at 0 degrees. The other curves in FIG. 1(a) are for an incidence angle of 0 degrees, and the other curves in FIG. 1(b) are for a distance r of 2d. The improvement in SNR corresponds to the area in the figure between the close-talking response and the farfield response. Note that the improvement is actually higher than can be seen in the figures due to the 1/r behavior of the sound pressure from a point source radiator. This effect is eliminated in the figure by normalizing the sound pressure in order to concentrate on the close-talking effect. It can be seen that the noise attenuation as well as the frequency response of the array depend highly on the distance and orientation of the close-taking array relative to the nearfield source.
Heinz Teutsch and Gary W. Elko, “An adaptive close-talking microphone array,” Proceedings of the WASSPA, New Paltz, N.Y., October 2001, the teachings of which are incorporated herein by reference, describe an adaptive method that estimates the distances and the orientation of a close-talking array based on time delay of arrival (TDOA) and relative signal level. The estimated parameters are used to generate a correction filter resulting in a flat frequency response for the close-talking array independent of array position. While this method provides a large improvement over conventional close-talking microphone arrays, it does not allow recovering the loss in attenuation of farfield sources due to orientation of the microphone array. As can be seen in FIG. 1(b), this loss can be significant. In addition, the array will become more sensitive to the orientation with increasing differential order as the main lobe becomes narrower.